J2EE testing procedures; need some help
Hi, Iam working on compiling some key items and write a small document on J2EE's best practices on Testing Procedures. This document may contain anywhere from basic to few complex testing procedures.
Iam not that familiar at the moment (i have very little experience doing J2EE testing) and iam trying to learn (through this document) about J2EE testing procedures and guidelines.
So, can somebody please give me some references or some ideas ?
Thanks a ton for all your help.
Hi, Iam working on compiling some key items and
write a small document on J2EE's best practices on
Testing Procedures. This document may contain
anywhere from basic to few complex testing
procedures. Sounds like your boss gave you an assignment that you're not up for. Have you ever done any testing? If not, how can you write about best practices?
I am not that familiar at the moment (i have very
little experience doing J2EE testing) and iam trying
to learn (through this document) about J2EE testing
procedures and guidelines.Do you have experience with Java EE? How do you test your stuff when you write it?
So, can somebody please give me some references or some ideas ?
Thanks a ton for all your help.Tell your boss I want the raise you get from my ideas.
Start with unit testing classes as you write them. Test Driven Development, when done properly, will be a big help. Look at TestNG and JUnit.
Mock objects can help here. If you've already tested your persistence layer, there's no need to worry about it for your service layer tests. Look at EasyMock.
Next is integration or business acceptance testing - combinations of classes and packages, front to back, entire use cases. You need to include the UI here. I'm looking at ThoughtWorks' Selenium and its IDE. It even pumps out Java code for you now. Very nice.
Stress testing is next. You'll need something that can simulate many users hitting the app at once. JMeter might help.
You have to worry about setting up test data in databases. It's got to stable while you're testing. It won't do to have tests break simply because data was changed. DbUnit might help.
Most of all, don't do all your testing at the end in one "big bang".
%
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Edited by: jim1768 on Mar 31, 2010 1:47 PM
Edited by: jim1768 on Mar 31, 2010 2:12 PM
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172.22.100.X VOICE
172.22.101.X DATA
my tftpserver = 172.22.1.150
i need some help configuring a sip trunk , i have 10 testing phonenumbers from vodafone , but i do not know where to start to get this working
i have tried
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml
but im getting stuck with what to fill in where .. is there anyone form NL whom has the same setup ? or similar ? or can give me some guidance on how to make the test callsok.. here goes
Building configuration...
Current configuration : 9721 bytes
! Last configuration change at 15:26:14 CET Thu Jan 2 2014
! NVRAM config last updated at 15:26:14 CET Thu Jan 2 2014
! NVRAM config last updated at 15:26:14 CET Thu Jan 2 2014
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
no aaa new-model
clock timezone CET 1 0
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-select 1 E1 0/0/0
network-clock-select 2 E1 0/0/1
dot11 syslog
ip source-route
ip cef
ip dhcp pool VOICE
network 172.22.100.0 255.255.255.0
option 150 ip 172.22.1.150
default-router 172.22.100.1
ip dhcp pool DATA
network 172.22.101.0 255.255.255.0
default-router 172.22.101.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
ip address trusted list
ipv4 172.22.1.50
ipv4 172.22.1.51
ipv4 172.22.100.1
ipv4 172.22.101.1
ipv4 62.140.159.225
callmonitor
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 3600 min 3600
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice register global
voice translation-rule 1
rule 1 /5123781291/ /601/
rule 2 /5123781290/ /600/
voice translation-rule 2
rule 1 /^112$/ /112/
voice translation-rule 3
rule 1 /^.*/ /0262610290/
voice translation-rule 4
rule 2 /600/ /5123788000/
rule 3 /601/ /5123788001/
rule 4 /^2(..)$/ /51237812\1/
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
voice translation-profile PSTN_CallForwarding
translate redirect-target 4
translate redirect-called 4
voice translation-profile PSTN_Outgoing
translate calling 3
translate called 2
translate redirect-target 4
translate redirect-called 4
voice-card 0
crypto pki token default removal timeout 0
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-31
controller E1 0/0/1
framing NO-CRC4
pri-group timeslots 1-31
ip ftp username cisco
ip ftp password cisco123
ip tftp source-interface GigabitEthernet0/0.1
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
no keepalive
interface GigabitEthernet0/0.1
encapsulation dot1Q 1 native
ip address 172.22.1.51 255.255.255.0
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 172.22.101.1 255.255.255.0
interface GigabitEthernet0/0.100
encapsulation dot1Q 100
ip address 172.22.100.1 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex full
speed 100
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http max-connections 16
ip http path flash:gui
ip route 0.0.0.0 0.0.0.0 172.22.1.50
tftp-server flash:7941/apps41.8-4-1-23.sbn alias apps41.8-4-1-23.sbn
tftp-server flash:7941/cnu41.8-4-1-23.sbn alias cnu41.8-4-1-23.sbn
tftp-server flash:7941/dsp41.8-4-1-23.sbn alias dsp41.8-4-1-23.sbn
tftp-server flash:7941/jar41sccp.8-4-1-23.sbn alias jar41sccp.8-4-1-23.sbn
tftp-server flash:7941/cvm41sccp.8-4-1-23.sbn alias cvm41sccp.8-4-1-23.sbn
tftp-server flash:7941/SCCP41.8-4-2S.loads alias SCCP41.8-4-2S.loads
tftp-server flash:7941/term41.default.loads alias term41.default.loads
tftp-server debug
control-plane
voice-port 0/0/0:15
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/0/1:15
voice-port 2/0/0
voice-port 2/0/1
voice-port 2/0/2
voice-port 2/0/3
voice-port 2/0/4
voice-port 2/0/5
voice-port 2/0/6
voice-port 2/0/7
voice-port 2/0/8
voice-port 2/0/9
voice-port 2/0/10
voice-port 2/0/11
voice-port 2/0/12
voice-port 2/0/13
voice-port 2/0/14
voice-port 2/0/15
voice-port 2/0/16
voice-port 2/0/17
voice-port 2/0/18
voice-port 2/0/19
voice-port 2/0/20
voice-port 2/0/21
voice-port 2/0/22
voice-port 2/0/23
mgcp profile default
dial-peer voice 1 voip
description **Incomming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
session protocol sipv2
session target ipv4:172.22.1.50
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9........
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 3 voip
description **Outgoing Call to SIP Trunk **
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[2-9]..[2-9]......
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[0-1][2-9]..[2-9]......
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 5 voip
description **911 Outgoing Call to SIP trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 911
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 6 voip
description **Emergency Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9911
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 7 voip
description **911/411 Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[2-9]11
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 8 voip
description **International Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9011T
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 9 voip
description **Star Code to SIP Trunk**
destination-pattern *..
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 10 voip
description **CUE Voicemail**
translation-profile outgoing PSTN_CallForwarding
destination-pattern 600
b2bua
session protocol sipv2
session target ipv4:172.22.1.155
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 11 voip
description **CUE Auto Attendant**
translation-profile outgoing PSTN_CallForwarding
destination-pattern 601
b2bua
session protocol sipv2
session target ipv4:172.22.1.155
dtmf-relay sip-notify
codec g711ulaw
no vad
sip-ua
authentication username 0262610290 password 7 15020A1F173D24362C realm 62.140.1
59.225
authentication username 0262610290 password 7 021605481811003348
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:62.140.159.225 expires 3600
sip-server ipv4:62.140.159.224
host-registrar
telephony-service
max-ephones 58
max-dn 192
ip source-address 172.22.100.1 port 2000
calling-number initiator
system message testing
cnf-file location TFTP tftp://172.22.1.150/
load 7960-7940 P00307020200.loads
load 7941 SCCP41.8-4-2S.loads
load 7941GE SCCP41.8-4-2S
time-format 24
dialplan-pattern 1 26261029.. extension-length 3 extension-pattern 9..
voicemail 600
max-conferences 12 gain -6
call-forward pattern 9.T
moh music-on-hold.au
web admin system name admin password password
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
directory entry 1 101 name 101
create cnf-files version-stamp 7960 Jan 02 2014 08:40:49
ephone-dn 1
number 290 secondary 0262610290
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 291 secondary 0262610291
name phone 2
hold-alert 30 originator
ephone-dn 3
number 292 secondary 0262610292
name Phone 3
hold-alert 30 originator
ephone-dn 4
number 293 secondary 0262610293
name Phone 4
hold-alert 30 originator
ephone-dn 5
number 294 secondary 0262610294
label Phone 5
hold-alert 30 originator
ephone 1
mac-address 0019.E88F.3BDD
button 1:1
ephone 2
mac-address 001E.4A92.0A27
type 7961
button 1:2
ephone 3
mac-address 0012.43F5.03AF
button 1:3
ephone 4
mac-address 000F.F7AC.502A
button 1:4
ephone 5
mac-address 0019.E851.090A
button 1:5
line con 0
line aux 0
line vty 0 4
login
transport input all
scheduler allocate 20000 1000
ntp master
end -
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